[asterisk-users] Asterisk call limitation
satish patel
satish_lx at hotmail.com
Tue Jun 21 08:44:39 CDT 2011
check this out: http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/258425
> From: kchehab at xplorium.com
> To: asterisk-users at lists.digium.com
> Date: Tue, 21 Jun 2011 13:25:39 +0300
> Subject: Re: [asterisk-users] Asterisk call limitation
>
> Any update ?
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Khaled W.
> Chehab
> Sent: Tuesday, June 21, 2011 12:40 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Asterisk call limitation
>
>
> The problem remains even when
>
> I add to /etc/init.d/asterisk
> ulimit -n 65536
>
> [root at localhost ~]# ulimit -a
> core file size (blocks, -c) 0
> data seg size (kbytes, -d) unlimited
> scheduling priority (-e) 0
> file size (blocks, -f) unlimited
> pending signals (-i) 65536
> max locked memory (kbytes, -l) 32
> max memory size (kbytes, -m) unlimited
> open files (-n) 1024
> pipe size (512 bytes, -p) 8
> POSIX message queues (bytes, -q) 819200
> real-time priority (-r) 0
> stack size (kbytes, -s) 10240
> cpu time (seconds, -t) unlimited
> max user processes (-u) 65536
> virtual memory (kbytes, -v) unlimited
> file locks (-x) unlimited
> [root at localhost ~]#
>
> -----Original Message-----
> From: Khaled W. Chehab [mailto:kchehab at xplorium.com]
> Sent: Tuesday, June 21, 2011 12:25 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Asterisk call limitation
>
> Can you please specify more
>
> 1-how to set the ulimit on
> [root at localhost ~]# ulimit
> unlimited
> [root at localhost ~]# ulimit --help
> -bash: ulimit: --: invalid option
> ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit]
> ---------------------------------------------------------------------
> How to set the ulimit command on in /etc/init.d/asterisk Since there is no
> parameter for ulimit in the file
>
> Thanks in advance
>
> Regards
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Satish Patel
> Sent: Tuesday, June 21, 2011 12:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk call limitation
>
> Oh! Wait you set ulimit for running shell You should set ulimit on
> asterisk. Also you can set ulimit command on asterisk startup file /
> etc/init.d/asterisk and restart asterisk also you can set in limit.conf file
>
> I had this issue before and I solved that way.
>
> --
> Sent from my iPhone
>
> On Jun 20, 2011, at 4:47 PM, "Khaled W. Chehab" <kchehab at xplorium.com>
> wrote:
>
> >
> > I tried the ulimit
> >
> > [root at localhost ~]# ulimit
> > Unlimited
> >
> > Then
> > sipp -sn uac -d 10000 -s 2005 127.0.0.1 -l 150
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:4 Up Playback(ss-
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s at from-trunk:5 Up SayAlpha(2005)
> >
> > 100 active channels
> > 100 active calls
> > 6407 calls processed
> >
> > [root at localhost ~]#
> > I find in /var/log/asterisk/full
> >
> > [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified
> > config file name '/etc/asterisk/extensions.ael'.
> > [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading
> > unistim.conf...
> > [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20
> > 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20
> > 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20
> > 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20
> > 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20
> > 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20
> > 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20
> > 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20
> > 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20
> > 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20
> > 16:44:26] WARNING[12908] file.c: Failed to write frame
> >
> > Khaled Chehab
> > NGN Eng.
> >
> >
> > Operations Office - Lebanon
> > Office : +961 1 868686 ext 115
> > Mobile: +961 3 045212
> > E-mail: kchehab at xplorium.com
> > MSN ID :KhalidChehab at hotmail.com
> > Web Site: http://www.xplorium.com
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Satish
> > Patel
> > Sent: Monday, June 20, 2011 11:24 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Asterisk call limitation
> >
> > It could be your OS limit try ulimit command.
> >
> > --
> > Sent from my iPhone
> >
> > On Jun 20, 2011, at 2:21 PM, "Kevin P. Fleming" <kpfleming at digium.com>
> > wrote:
> >
> >> On 06/20/2011 01:09 PM, Khaled W. Chehab wrote:
> >>> Dears,
> >>>
> >>>
> >>>
> >>> i am using sipp to test asterisk(1.6.22) performance ,but when i
> >>> limit the calls to 150 ,only 100 active calls on asterisk found ?why
> >>>
> >>> sipp -sn uac -d 10000 -s 2005 127.0.0.1 -l 150
> >>
> >> You did not provide any log output, or anything that could be used to
> >> try to help you understand your problem. Without any details, any
> >> reply you get would be just a guess, nothing more.
> >>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> Regards
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> Khaled Chehab
> >>>
> >>> NGN Eng.
> >>>
> >>>
> >>>
> >>> Description: xplorium
> >>>
> >>> Operations Office - Lebanon
> >>>
> >>> Office : +961 1 868686 ext 115
> >>>
> >>> Mobile: +961 3 045212
> >>>
> >>> E-mail:<mailto:kchehab at xplorium.com> kchehab at xplorium.com
> >>>
> >>> MSN ID :KhalidChehab at hotmail.com
> >>>
> >>> Web Site: http://www.xplorium.com
> >>
> >> Please refrain from including 20-line signature blocks in your
> >> messages to the Asterisk mailing lists (or really, anywhere). Your
> >> message had three lines of content and 30+ lines of non-content.
> >>
> >> --
> >> Kevin P. Fleming
> >> Digium, Inc. | Director of Software Technologies
> >> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype:
> >> kpfleming
> >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
> >> www.digium.com & www.asterisk.org
> >>
> >> --
> >> _____________________________________________________________________
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> >> --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >> http://www.asterisk.org/hello
> >>
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> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> > --
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> > -- New to
> > Asterisk? Join us for a live introductory webinar every Thurs:
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> >
> >
> > --
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> >
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> >
>
> --
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> --
> _____________________________________________________________________
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